Certain embodiments of the present invention relate to voice processing for broadband communication systems. More specifically, certain embodiments relate to a method and system for an adaptive multimode media queue for supporting data sampled at different rates.
Packet based telephony such as Internet Protocol (IP) telephony may provide an alternative to conventional circuit switched telephony, the latter of which may typically require the establishment of an end-to-end communication path prior to the transmission of information. In particular, IP telephony permits packetization, prioritization and simultaneous transmission of voice traffic and data without requiring the establishment of an end-to-end communication path. IP telephony systems may capitalize on voice-over-packet (VoP) technologies, which may provide a means by which voice, video and data traffic may be simultaneously transmitted across packet networks. The data may include video data.
Voice quality (VQ) may define a qualitative and/or quantitative measure regarding the quality and/or condition of a received voice signal. Voice clarity may be an indicator of the quality or condition of a voice signal. Voice quality may be an important parameter that may ultimately dictate a quality of service (QOS) offered by a network service provider. The following factors, for example, may affect the voice quality and/or condition of a voice signal—noise, echo, and delay or packet latency. However, the effects of these factors may be cumulative. In this regard, factors such as delay and latency may exacerbate the effects of echo. Delays that may affect the voice quality may include, but are not limited to, routing, queuing and processing delays.
Various VoP specifications, recommendations and standards have been created to ensure interoperability between various network components, and to create an acceptable QOS which may include voice quality. For example, the International Telecommunications Union (ITU) ratified H.323 specification, which may define the processes by which voice, video and data may be transported over IP networks for use in VoP networks. H.323 addresses, for example, delay by providing a prioritization scheme in which delay-sensitive traffic may be given processing priority over less delay-sensitive traffic. For example, voice and video may be given priority over other forms of data traffic.
H.323 also addresses voice quality by specifying the audio and video coders/decoders (CODECs) that may be utilized for processing a media stream. A CODEC may be a signal processor such as a digital signal processor (DSP) that may be adapted to convert an analog voice and/or video signal into a digital media stream and for converting a digital media stream into an analog voice and/or video signal. In this regard, a coder or encoder portion of the CODEC may convert an analog voice and/or video signal into a digital media stream. Additionally, a decoder portion of the CODEC may convert a digital media stream into an analog voice and/or video signal. Regarding the CODEC for audio signals, H.323 may support recommendations such as ITU-T G.711, G.722, G.723.1, G.728 and G.729 recommendations. ITU-T G.711 may support audio coding at 64 kbps, G.722 may support audio coding at 64 kbps, 56 kbps and 48 kbps, G.723.1 may support audio coding at 5.3 Kbps and 6.3 Kbps, G.728 may support audio coding at 16 kbps and G.729 may support audio coding at 8 kbps.
The voice quality of a speech CODEC may be dependent on factors such as the type of encoding and/or decoding algorithm utilized by the CODEC. In general, some CODECs may utilize compression algorithms that remove redundant information from the analog signal. Such compression algorithms may permit at least a close replication of an original analog signal. In this case, the bandwidth required for transmitting any resultant signal may be reduced. Other CODECs may utilize algorithms that analyze the signal and retain only those portions that are deemed to be of cognitive importance. These algorithms may reproduce a close approximation to the original signal. Notwithstanding, in this latter case, bandwidth utilization may be superior to the former case where redundant information may be removed. Accordingly, depending on application requirements and hardware limitations, one or more algorithms may be utilized to optimize performance.
Moreover, although economic attractiveness of VoP has lured network access providers and network transport providers away from traditional circuit switching networks, factors such as the extensiveness of embedded legacy systems and customer demands, for example, have dictated the coexistence of both packet switched and circuit switch networks. Accordingly, new technologies and techniques such as audio and video coding and decoding may be required to support various modes of operation utilized by each system.
Traditional voice telephony products are band-limited to 4 kHz bandwidth with 8 kHz sampling, known as “narrowband”. These products include the telephone, data modems, and fax machines. Newer products aiming to achieve higher voice quality have doubled the sampling rate to 16 kHz to encompass a larger 8 kHz bandwidth, which is also known as “wideband” capable. The software implications of doubling the sampling rate are significant. Doubling the sampling rate not only requires doubling the processing cycles, but nearly doubling the memory used to store the data.
Doubling memory and processor cycles requirements is expensive because the memory and processing power footprints of DSPs are generally small. Implementing wideband support thus requires creativeness to optimize both memory and cycles.
Additionally, much of the software providing various functions and services, such as echo cancellation, dual-tone multi-frequency (DTMF) detection and generation, and call discrimination (between voice and facsimile transmission, for example), are written for only narrowband signals. Either software must be written for wideband signals, or the wideband signal down-sampled. Where the software is modified, the software should also be capable of integration with preexisting narrowband devices. Providing software for operation with both narrowband and wideband devices is complex and costly.
Accordingly, it would be advantageous if a scheme for seamlessly integrating services for both narrowband devices and wideband devices were provided.
Further limitations and disadvantages of conventional and traditional approaches will become apparent to one of skill in the art, through comparison of such systems with some aspects of the present invention as set forth in the remainder of the present application with reference to the drawings.